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gerrit-public.fairphone.software / platform / external / webrtc / 91811e2b0457e091886508894a771f0e12054d0b
  1. 91811e2 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
  2. a4c5abb Make sure padding packets are sent. by stefan@webrtc.org · 11 years ago
  3. bb25256 Added changes in apprtc demo to ignore turn address through query string for FF. Also made sure that the iceServers array doesnot include transport parameter in turn url for FF. Finally removed turn: from the turn_url when creating iceservers for pre-M28 chrome. by vikasmarwaha@webrtc.org · 11 years ago
  4. 3348ae2 mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function. by sergeyu@chromium.org · 11 years ago
  5. bb4f225 Roll libvpx to 207593. -pick up libvpx roll to c259af4f. by marpan@webrtc.org · 11 years ago
  6. 6eb53f7 Fix memory bot failure by hclam@chromium.org · 11 years ago
  7. 2e402ce Enqueue packet in pacer if sending fails by hclam@chromium.org · 11 years ago
  8. 9ca7360 VCM: removing max jitter estimate by mikhal@webrtc.org · 11 years ago
  9. 0851df8 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. by andrew@webrtc.org · 11 years ago
  10. 8ccb9f9 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 11 years ago
  11. 2d7617a Add dummy Android test APK to be used for buildbot automation testing. by kjellander@webrtc.org · 11 years ago
  12. d7148c8 Use 3 threads for higher than 720p resolutions by fbarchard@google.com · 11 years ago
  13. 30fb7b8 Add a log message to see video delay break down by hclam@chromium.org · 11 years ago
  14. 6cfe178 Chromium Android tools for test execution. by kjellander@webrtc.org · 11 years ago
  15. a20eb91 Make ScreenCapturerMac work in versions of OSX before Lion. by sergeyu@chromium.org · 11 years ago
  16. 9e18279 Enable ScreenCapturer unittests by sergeyu@chromium.org · 11 years ago
  17. a590b41 Use intptr_t to represent window IDs on all platforms. by sergeyu@chromium.org · 11 years ago
  18. 508a84b Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
  19. 50fb4af Revert r4145 "Revert 4127 "Switch frame list implementation to std::map."" by stefan@webrtc.org · 11 years ago
  20. c8b29a2 Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de..."" by stefan@webrtc.org · 11 years ago
  21. 7262ad1 Fix AV sync issue by hclam@chromium.org · 11 years ago
  22. 9b23ecb Log current and target AV delay in ViESyncModule by hclam@chromium.org · 11 years ago
  23. 63e9888 Merge more tests into modules_{unit,integration}tests. by kjellander@webrtc.org · 11 years ago
  24. f27389c WebRTCDemo: ensures that using front and back camera work as expected. by henrike@webrtc.org · 11 years ago
  25. d4ed1a3 Fixes linker issue with no op trace. by henrike@webrtc.org · 11 years ago
  26. a193339 Apprtc CSS: Add flip to local view of FireFox and remove warning of Canary by braveyao@webrtc.org · 11 years ago
  27. fee739c Risk of division by zero. by turaj@webrtc.org · 11 years ago
  28. dd97ef4 Revert 4211 "Build all java files into jar for each module on An..." by fischman@webrtc.org · 11 years ago
  29. 20a993f Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  30. 935d705 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  31. 04996cd Fix breakage due to test_fec conversion to gtest. by kjellander@webrtc.org · 11 years ago
  32. 22bbbdf Convert test_fec to gtest by kjellander@webrtc.org · 11 years ago
  33. 7124dd8 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  34. 18275a8 Update bots to make LKGR progress. by kjellander@webrtc.org · 11 years ago
  35. b097670 G722_1/G722_1C codecs won't instantiate by tina.legrand@webrtc.org · 11 years ago
  36. 2ef9513 libyuv r723 with convert util -attenuate feature used to fix transparent pixels used by Effects. By attenuating and then unattenuating, any transparent pixels will have RGB value of black, which will filter correctly when bilinear resized. by fbarchard@google.com · 11 years ago
  37. 6c35e0b Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
  38. 6d6d95e Add support for test disable files in webrtc_tests.py by kjellander@webrtc.org · 11 years ago
  39. 1374965 Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
  40. 4af0878 Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows). by alexeypa@chromium.org · 11 years ago
  41. 5e03f8a Landing binary cursor image files to be used in a follow up CL. by alexeypa@chromium.org · 11 years ago
  42. dfa1c4a libyuv r722 for OWNERS file for chromium, white space fix for lint, unittests on scale use randomize to reduce overhead, and neon change from vld1.u8 to vld1.8 for better compiler portability. by fbarchard@google.com · 11 years ago
  43. fe6b571 AppRTCDemo: delete hosted android_channel.html now that it's no longer necessary. by fischman@webrtc.org · 11 years ago
  44. 5137b97 Updated WebRTC version to 3.33 by elham@webrtc.org · 11 years ago
  45. 509754c Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
  46. 1819fd7 RW lock access to ssrc maps in VideoCall. by pbos@webrtc.org · 11 years ago
  47. adb51f5 Add back the WEBRTC_DIRECT_TRACE flag. by solenberg@webrtc.org · 11 years ago
  48. 83a062c AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout() by braveyao@webrtc.org · 11 years ago
  49. 569fdef Revert some variables to uint32_t to fix compile errors on Mac gcc. by andrew@webrtc.org · 11 years ago
  50. 6f69eb7 Allow audio devices with up to 64 channels on Mac. by andrew@webrtc.org · 11 years ago
  51. 1064cf0 Fixed Rtp/Rtcp tests by pwestin@webrtc.org · 11 years ago
  52. 6367fe8 Fix relative path to .gitignore and other minor changes. by andrew@webrtc.org · 11 years ago
  53. 3ba883f Removing functionality for inserting pre-encoded frames instead of raw by mflodman@webrtc.org · 11 years ago
  54. b69cc15 Add script for appending entries to .gitignore. by andrew@webrtc.org · 11 years ago
  55. da71044 Fix size_t to int conversion error on Win64. by andrew@webrtc.org · 11 years ago
  56. 7e4ff35 Remove fake screen capturer because it's not used anywhere. by sergeyu@chromium.org · 11 years ago
  57. 8d80fa8 Fix for STL vector function data not available. by pwestin@webrtc.org · 11 years ago
  58. d30859e Connect ACM with RTP module for audio NACK. by pwestin@webrtc.org · 11 years ago
  59. a305e96 Nack for audio. by turaj@webrtc.org · 11 years ago
  60. d9c4658 Fix leaks in DesktopRegion by sergeyu@chromium.org · 11 years ago
  61. 2b3a29a Implement DetectNumberOfCores on Android and make it consistent on Linux and Android by fischman@webrtc.org · 11 years ago
  62. db24995 Wire up Nack for Voe by pwestin@webrtc.org · 11 years ago
  63. 7f1b0ae Fix init list for VideoSendStream::Config::Rtp. by pbos@webrtc.org · 11 years ago
  64. 025f4f1 Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 11 years ago
  65. fec34d7 Merge webrtc_utility_unittests into modules_unittests. by kjellander@webrtc.org · 11 years ago
  66. b2d29bd Restore relative include paths to libyuv. by andrew@webrtc.org · 11 years ago
  67. 3942f3a Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer. by turaj@webrtc.org · 11 years ago
  68. 16d78bd Fix scale.cc build error with mingw64 -m32 gcc by fbarchard@google.com · 11 years ago
  69. 9238de9 resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero. by turaj@webrtc.org · 11 years ago
  70. 3d34f66 Move screen capturers from chromium to webrtc. by sergeyu@chromium.org · 11 years ago
  71. b7a8f43 Roll chromium_revision in webrtc 199267:203806 by fischman@webrtc.org · 11 years ago
  72. 430464c Add WebKit/Tools/Scripts to support Android test execution. by kjellander@webrtc.org · 11 years ago
  73. a817962 Refactor padding and rtp header functionality. by stefan@webrtc.org · 11 years ago
  74. de98478 Update the remote bitrate estimator before passing the packet to the RTP module. by stefan@webrtc.org · 11 years ago
  75. 6998c8e Remove XvRenderer. by pbos@webrtc.org · 11 years ago
  76. 8ad3ec9 Fix build error introduced with r4168. by stefan@webrtc.org · 11 years ago
  77. c3cc375 Add support for padding in pacer. by stefan@webrtc.org · 11 years ago
  78. c69ae69 Include files from webrtc/.. paths in common_video/ by pbos@webrtc.org · 11 years ago
  79. ba7f6a8 Include files from webrtc/.. paths in tools/ by pbos@webrtc.org · 11 years ago
  80. 5156c94 Disable neteq_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
  81. b6e49aa Disable audio_decoder_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
  82. 6eba277 Disable audio_coding_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
  83. e001b57 Do not hold a lock when calling VCMReceiveCallback::FrameToRender. by fischman@webrtc.org · 11 years ago
  84. 3ee13e4 Optimized DesktopRegion implementation. by sergeyu@chromium.org · 11 years ago
  85. 34a7735 Removed unused class members to enable clang=1 android build. by fischman@webrtc.org · 11 years ago
  86. 6eb0f6a Setting SSRC in vie_loopback_test by mikhal@webrtc.org · 11 years ago
  87. 0a38432 Fix error in mixing test for supported sample rates. by andrew@webrtc.org · 11 years ago
  88. fa64a59 Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 11 years ago
  89. c1eb560 Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 11 years ago
  90. 31c5f1c Remove ancient and unused CNG test. by andrew@webrtc.org · 11 years ago
  91. 2b3a865 Revert 4149 "bug fixes for extremely large images - 10000x10000 ..." by mikhal@webrtc.org · 11 years ago
  92. b35d2e3 Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 11 years ago
  93. b1bba16 Prevent excessive logging in jitter buffer by hclam@chromium.org · 11 years ago
  94. 85f2865 bug fixes for extremely large images - 10000x10000 and 100000 pixel wide. by fbarchard@google.com · 11 years ago
  95. a6494e6 roll libyuv to r711 for scaler fix to webrtc unittests that scale up and down and check for fairly similar results. by fbarchard@google.com · 11 years ago
  96. 694cdc6 Revert 4104 "Refactor jitter buffer to use separate lists for de..." by tnakamura@webrtc.org · 11 years ago
  97. 4d9c07a Revert 4127 "Switch frame list implementation to std::map." by tnakamura@webrtc.org · 11 years ago
  98. 5ed7051 Apprtc: not to start the call until we get Turn response. by braveyao@webrtc.org · 11 years ago
  99. f9f39d5 Add a drover.properties file for reference. by andrew@webrtc.org · 11 years ago
  100. eed919d MIPS optimizations for the following functions: by andrew@webrtc.org · 11 years ago